On Fri, Dec 30, 2005 at 11:47:41PM -0500, Ross Vandegrift wrote: > On Fri, Dec 30, 2005 at 05:42:32PM -0500, Lee Revell wrote: > > LDAS (the low delay audio streamer) does something similar - it has to > > compensate for the drift between two machines clocks' that exchange > > audio over the network but you could apply the algorithm to two cards on > > one box. It uses the ALSA API and is believed to accomplish this with > > the lowest achievable latency. > > The site sounds like it mostly attempts to compensate for network latency, > not clock drift between the hosts, though the PDF mentions some > "watermark algorithm" (google didn't turn up anything relevant to > timekeeping devices).
Actually, we also try to compensate for clock drift, which _is_ a problem. This is done by watching the (low pass filtered) length of the receiver buffer. If the buffer grows, it is assumed that the receivers sound card clock is slower than the senders, and vice versa. The current solution is not very elegant, single samples are dropped or reused as appropriate to keep the receiver buffer length withing given bounds (the "watermarks"). A better solution, which I would like to implement, is to use an (adaptive?) sample rate converter. Normally, an SRC between two almost alike sampling frequencies (as we have here) would demand long filters (and high latency). However, professor Tor Ramstad at NTNU has written an article about how such SRCs may be implemented in more efficient ways. Asbjørn ------------------------------------------------------- This SF.net email is sponsored by: Splunk Inc. Do you grep through log files for problems? Stop! Download the new AJAX search engine that makes searching your log files as easy as surfing the web. DOWNLOAD SPLUNK! http://ads.osdn.com/?ad_idv37&alloc_id865&op=click _______________________________________________ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
